In a mobile communications service, due to a packet loss and delay variation on a network, it is inevitable to cause a frame loss, resulting in that some speech/audio signals cannot be reconstructed using a decoded parameter and can be reconstructed only using a frame erasure concealment (FEC) technology. However, in a case of a high packet loss rate, if only the FEC technology at a decoder side is used, a speech/audio signal that is output is of relatively poor quality and cannot meet the need of high quality communication.
To better resolve a quality degradation problem caused by a speech/audio frame loss, a redundancy encoding algorithm is generated. At an encoder side, in addition to that a particular bit rate is used to encode information about a current frame, a lower bit rate is used to encode information about another frame than the current frame, and a bitstream at a lower bit rate is used as redundant bitstream information and transmitted to a decoder side together with a bitstream of the information about the current frame. At the decoder side, when the current frame is lost, if a jitter buffer or a received bitstream stores the redundant bitstream information containing the current frame, the current frame can be reconstructed according to the redundant bitstream information in order to improve quality of a speech/audio signal that is reconstructed. The current frame is reconstructed based on the FEC technology only when there is no redundant bitstream information of the current frame.
It can be known from the above that, in the existing redundancy encoding algorithm, redundant bitstream information is obtained by means of encoding using a lower bit rate, and therefore, signal instability may be caused, resulting in that quality of a speech/audio signal that is output is not high.